The present invention relates to a speech signal coder for coding a speech signal of speech, music and so forth, and more particularly, to a signal coder capable of permitting high quality coding at low bit rate quantization.
Methods of efficiently coding a speech signal spectrum on a frequency axis are well known in the art as disclosed in, for instance, T. Moriya, "Transform coding of speech using a weighted vector quantizer" and N. Iwakami, "High-quality audio-coding at less than 64 kbit/s using transform-domain weighted interleave vector quantization (TWINVQ)".
In these methods, DCT (Discrete Cosine Transform) coefficients of the speech signal are obtained by making an orthogonal transform thereof based on DCT for a number N of different points.
The DCT coefficient are then m divided at a number (M.ltoreq.N) of points. The speech signal is then vector quantized by making a codebook retrieval for each of the M division points.
However, these prior art signal coders had the following problems in the speech signal coding.
Firstly, DCT coefficients of N points are all quantized uniformly. Therefore, reducing the bit number of a vector quantizer to reduce the bit rate, leads to the difficulty of obtaining satisfactory DCT coefficients which have a perceptually important role. In other words, although relatively satisfactory speech quality is obtainable by high bit rate coding, reducing the bit rate leads to extreme deterioration of the speech signal quality.
A second problem is posed by increasing the number M of points of the DCT coefficient division to improve the efficiency of vector quantization. Increasing the number M of points of the DCT coefficient division results in an increase of the dimension number of the vector quantizer. The dimension number exponentially increases the computational effort necessary for the vector quantization, and makes it impossible to reduce the bit rate.